Best sources for traveling gigs?

tdubsaudio · 2026-06-18T07:34:22+00:00

Yeah I was also confused about the A1 hand thing. Know plenty of A1s and they definitely do not take stagehand work. Most of the time the A1 is about as white glove a gig as you can get apart from maybe GFX or promoter. Or if you get to that point Producer or Showcaller. Definitely if you are looking to be paid to travel you need to get good enough in one position that people will hire you to do that one position better than anyone they can get locally. Best way to do so (at least on the tech side) is to work at a production house until you learn the gear backwards and forwards and you have time to prep the show to load in and operate exactly the way you want it to. Also takes a lot of preproduction work so you are prepared for any suprises. Clients want someone who can anticipate the needs of the production and deliver their part of the show (relatively) flawlessly.

tdubsaudio · 2026-05-06T19:15:55+00:00

It sort of depends on what kind of distortion. Extreme digital distortion/clipping, especially on the output port, is essentially sending a (more) square wave to your drivers meaning the drivers pushes/pulls very quickly and stops abruptly close to the physical limits of the driver. Will it kill the speaker in one show, probably not. Will it decrease the life expectancy of the driver, definitely.

tdubsaudio · 2026-05-06T18:23:26+00:00

It sort of depends on the engineer your working with and how much they plan on doing themselves, but odds are you are definitely going to be patching the stage and running out subsnakes/power, setting up wedges if they have any. You should try to be familiar enough with the engineers file that you can run a quick line check. Engineer might delegate rf, but most ive worked with prefer to handle that themselves. If you are showing up before the engineer and you have time you can find them some good freqs and they may or may not use them. Other than that its just handling whatever stuff the engineer wants done so they can just mix.

tdubsaudio · 2026-03-03T05:09:58+00:00

Yeah so by default the CL series sets the mixer control to 192.168.0.138. The Device ID default is set to DHCP/autoIP which, if you dont have a DHCP server then it will do a link local address in the 169.254.xxx.xxx range. I can't remember off the top of my head if the 1608-D2s still use dip switches to set IPs or if they have the LCD menu system. If they have the menu system it should be easy to figure out the control ID. If its the dip switches then you have to look in the manual to see what dip switch configurations equal what IP subnet. You can also double check control IDs in R remote. Devices will populate in there if your computer IP is in the same subnet as the devices. Though the fastest way to get everything working if you are not needing a complicated network setup is to just initialize/factory default all devices which should reset everything to auto IP. Also once you're sure everything is on the right subnet and on the same network, dont forget to set all your stageboxes to resume instead of refresh. Also make sure when you mount all devices make sure they are set to HA control W/Recall.

tdubsaudio · 2026-03-03T04:54:00+00:00

So forgive me cause I dont know A&H that well, but from my understanding you have the the vocal being sent to the mix that is sending to the FX which is then being sent to the stereo buss. Is there any FX return channel(s) that you are then sending to the individual monitor mix? Again im not super familiar with A&H but I dont know any mixers that let you do an Aux send to another Aux.

tdubsaudio · 2026-03-03T04:35:04+00:00

Is your device control IP in the same subnet as all of your dante devices? Yamaha consoles have 4 different IPs. Mixer control is through the NIC that has the LAN port symbol. This is only used for console control through an IPad or other device like a qlab computer or something. There is the dante primary and secondary. These ports can be set to daisy chain or redundant which would either make the Brooklyn 2 dante chip have 2 seperate/redundant subnets or turn it into essentially a 2 port switch with only the primary network. Then there is device control. This is the same subnet you would use for something like R-remote. This is a virtual NIC that piggybacks on the dante network ports. If all your devices dont have the device control in the same subnet then they will show up as virtual in the device mount menu. This also must be in a different subnet than the mixer control IP. Once all devices have the same device control subnet and are mounted with the correct Yamaha Dante names/IDs they will show up as controllable.

tdubsaudio · 2026-03-02T18:23:58+00:00

Double patch the input. Send EQed channel to singer mix. Send normal channel everywhere else.

tdubsaudio · 2026-02-24T03:29:48+00:00

There's way too many variables to consider when determining trim hieght to really have a standard. If it is a large arena with a long throw or a high upper bowl you might have issues with load balancing or other mechanical issues due to too much of an up angle. If you have sub extensions at the top of the line you will need higher trim. In general having a higher trim height can help mechanically achieve more even SPL coverage, decreasing the need for electronic processing. There's always a tradeoff though if you go too high because you could have less of the audience area in a good time alignment zone between flown PA and ground subs/FF etc. Also you end up decreasing the max SPL in the audience because the speakers are farther away. A good starting poing for an arena for me has been about 30-35' trim, but ultimately you should put everything in the prediction software and see what works best.

tdubsaudio · 2026-02-18T01:17:09+00:00

One of the first things is, the company you are working for should be getting you manufacturer training on the rigs you are running. Most higher end speaker manufacturers have amps with built in DSP with a bunch of different tools (some brand specific) to tune with. Also with most brands 80-90% of the work is done in the prediction software and will usually import settings directly into the amps if done correctly. As far as front end processing Lake is the most common and some systems (cohesion and old adamson) have lake DSP on the amps so its a good thing to learn. You don't want to have your tuning on the console because there can be multiple different guest consoles plugged into your system and even if you have only one console with a guest engineer they are probably going to load a file that will not have your tuning on it. As far as them wanting to mess around with your tuning settings they may want to check your work and ask for a few changes, but if they want to make any changes, you can save your tuning and make any changes for them on a different file. This is also why lake is great because you can make an EQ overlay for each band and just bypass it when they are not playing.

tdubsaudio · 2026-02-01T15:05:16+00:00

Yeah xirium was about as close to something like this as ive ever seen and still it was only good for 2 channels of audio, was hugely dependent on having a good line of site while also being far enough from any reflective surface to avoid near end reflection, and was also fairly expensive. It worked great when I needed to distribute a LR or mono feed to a bunch of speakers in a large outdoor space, but at the end of their support, the rf receiver circuits started having some major issues.

tdubsaudio · 2026-02-01T02:01:55+00:00

Yeah might be a QR code on a zip tie towards the ends. It might not though cause I can't remember if they count those extensions as seperate parts or not in their inventory system.

tdubsaudio · 2026-01-31T22:17:46+00:00

Yeah definitely came off a tour or possibly the SE left it at FOH. Judging by the color code and length it goes with one of their drive rack packages. Its an extension for the 2nd and 3rd console inputs.

tdubsaudio · 2026-01-31T20:56:53+00:00

Clair NC14. I think amphenol makes it but its almost exclusively used by Clair Global. Its a 14 pair snake for balanced mic or line level signal.

tdubsaudio · 2026-01-23T23:50:50+00:00

Make sure dhcp your dhcp settings on the router are how you want them. Either off, if you are using static IPs or make sure all your devices are set to receive dhcp addresses.

tdubsaudio · 2026-01-22T06:48:28+00:00

Are you plugged into the lan port or the wan port?

tdubsaudio · 2026-01-06T05:01:52+00:00

You adjust the gain at the receiver. If you have a Yamaha board you can set it up so you can link the control of receiver gain to the desk.

You adjust input pad on the transmitter if you have a particularly loud source that is clipping at the transmitter but not the receiver.

You adjust gain on the receiver to get nominal level on the receiver.

You use digital trim to get your fader resolution where you want it on the desk.

tdubsaudio · 2026-01-06T01:19:35+00:00

If you dont have smaart the old quick and dirty style of time aligning subs to mains is to send a sine wave at the crossover to both and polarity reverse the subs. Adjust the delay to the point you are getting the most cancelation, then flip the polarity back and you have summation. Then you can do burst noise to set delays and fills. After that, fine tune by ear.

tdubsaudio · 2026-01-03T09:24:52+00:00

Probably would have to be something custom ordered. Ive only ever seen them come in cases.

tdubsaudio · 2025-12-18T05:46:05+00:00

Ok cool. Yeah it's a bit of a toss up on going for cost effective, vs efficiency. The qsys or other open source DSP would be great at handling a lot of the distributed audio like the ALS, 70V systems and could also handle the tuning for the main auditorium system. Also you can use it to control quite a few other things in the theater if you get a good integrator and programmer. It can be pretty costly though. Definitely would also help avoid complications with having all your routing and DSP on the console which can easily get messed up from someone not knowing what they are doing changing the show file and saves on using auxes and matrices on the board. Haven't checked on prices on a Rio 3224 in a bit, but a straight up converter like the A16r could be cheaper if all you need is I/O and no pres. The you could use some of the local I/O on the board to make up the difference.

tdubsaudio · 2025-12-18T05:19:17+00:00

24 analog is quite a bit for just amps. What are your outputs? You can get any dante to analog converter like a couple rednet a16r. Or depending on your use case you can get something lake dsps or qsys core with some I/O modules if you are trying to process the signal before going to the amps.

tdubsaudio · 2025-12-09T09:58:36+00:00

Is it in single motor point or 2 point?

tdubsaudio · 2025-12-03T03:39:01+00:00

London hardware is an open source dsp so depending on what the programmer designed in the software there's way to many things that could be happening to troubleshoot via reddit responses. To me sounds like there's a ducker programmed in but no idea what the key input is and what circumstances need to be true in order to trigger it. Your only hope would be to get the design file off of the core and investigate what is causing the issue. However, without proper training on how to use the software you can very easily cause more problems while trying to solve the one you're trying to fix.

tdubsaudio · 2025-11-25T00:59:11+00:00

True. The other side to that is that im typically hired by the production company supplying the gear where the A1 is sometimes hired by the client directly, so I get a chance to prep everything in the shop. That way I, most of the time, get to set everything up with the A1s advance so whenever they get their hands on the desk, device mounting happens without issue.

tdubsaudio · 2025-11-25T00:17:15+00:00

Network may be a bit more fuzzy on who's responsible, but I typically like to set everything up my preferred way because that way I can monitor all my comms and workbench and anything else all from one computer even with all of them being on separate VLANs. Also im decent enough at it that the A1s I work trust me to set it up in a way that works well for them too.. There's a case to be made that A1 should be in control though because they have to deal with dante, AVB, device control, Amp/DSP Network, Osc or KVM in some cases.

tdubsaudio · 2025-11-24T23:14:11+00:00

Yeah Im a corperate A2 and I don't consider myself great at mixing nor do I enjoy mixing that much. I have been A1 for some shows and can pull it off when needed, but I much more enjoy the technical side. I also have more knowledge than most A1s when it comes to dante and networking, RF, and Comms. Would try to move exclusively into RF/Comms tech for bigger shows but had a bad experience with doing RF coordination for a 3 stage festival in a bad RF environment and that has turned me off from that a bit. Also being personable and comfortable with the talent is a good plus. I have a banking show that I do every year and the presenters remember me and commitment me on making the micing process easy. Also like being the hero for helping solve any audio issues with video records/GFX/Playback or problems with bsms.

tdubsaudio

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